The present invention relates to voice communications and more particularly to packetized voice communications transferred over an Internet Protocol (IP) packet network.
The traditional telephone network, known as the Public Switched Telephone Network (PSTN) is a vast network that carries voice traffic from phone to phone around the world. The PSTN is a circuit switched network which uses an array of switches to form a dedicated line connection extending between the phones for the duration of the call.
Packet networks operate differently than circuit switched networks, breaking up the data or voice traffic into small packets or datagrams which are sent independently across the packet network. A dedicated line is not established between endpoints in a packet network and the separate packets may travel different routes through the network to reach the destination.
Voice traffic can also be sent from phone to phone using a combination of both packet networks and the PSTN. Service providers effectively utilize the benefits of both networks by providing an intermediate managed network 10 shown in FIG. 1 which connects customers to both the PSTN 12 and a global packet network 14, such as the Internet. Customers can have one or more enterprises 15 each having a private network 16 connected to the managed network 10. Each enterprise can include a plurality of endpoints 18 which may be phones, computers, software controlled phones called softphones or any other known endpoints.
The managed network 10 offers customer enterprises 15 a variety of voice and data services at lower costs. For example, toll charges associated with establishing a dedicated line connection can be avoided using a packet network. Also, compression techniques enable packetized voice traffic to be transferred over the PSTN 12 using less bandwidth than typical PCM voice signals.
To move voice traffic over packet networks 10, 14, 16, voice conversations are digitized and packetized. The voice packets are identified for proper routing over the packet network using a known packetization format generally known as Voice over Internet Protocol (VoIP). VoIP uses IP addressing schemes to uniquely identify the source and destination endpoint addresses.
Public IP addresses are unique addresses on the global IP network. However, there are a limited number of unique public IP addresses available according to the IP address format defined by Request for Comments (RFC) 791 (Internet Architecture Board). In order to conserve IP addresses, enterprises 15 which administer their own private networks 16 can use private IP addresses. Separate private networks 16 can use the same private addresses. The private addresses uniquely identify the endpoints within the private network, but are not unique to the global IP packet network 14 and perhaps the managed network 10.
However, to interconnect these private networks 16 address resolution is needed to eliminate addressing conflicts since endpoints 18 from different enterprises 15 may be using the same IP address. Network Address Translation (NAT) has been used for data traffic such as emails, web browsing, etc. to translate between private and public IP addresses to enable private and public networks to be interconnected.
VoIP presents new challenges for NAT, since VoIP traffic packets have IP addresses embedded in the payloads of the packet envelope. Previously, VoIP NAT has been done using a dedicated NAT device, such as a router or firewall 19, located at each enterprise site. However this approach becomes increasingly more difficult and costly to implement as more private networks 16 are serviced by the managed IP network 10 and as more VoIP protocols are implemented in the industry.
Accordingly, it is desirable to provide VoIP NAT which is scalable and less costly to implement for a large number of private networks 16 connected to intermediate networks such as those managed by service providers.